Communication bandwidth is becoming an increasingly valuable commodity. Media signals, including video and audio signals, may consume enormous amounts of bandwidth depending on the desired transmission quality. Data compression is therefore playing a correspondingly important role in communication.
Generally, the sending party selects a codec (compressor/decompressor) for compressing and decompressing media signals. A wide variety of codecs are available. General classifications of codecs include discrete cosine transfer (DCT) codecs, fractal codecs, and wavelet codecs.
The sending party will also typically decide on various codec settings that will apply throughout the communication session. Because the codec settings affect the “quality” of the transmission, i.e., how similar a received and decompressed signal is to the original, such settings are often referred to as quality settings.
In general, quality settings affect the amount of bandwidth required for the transmission. Higher quality settings typically consume greater bandwidth, while lower quality settings require lesser bandwidth.
Unfortunately, the bandwidth required for sending each frame of a media signal is variable, as is the overall amount of available bandwidth. Using a single set of quality settings throughout a transmission does not take into account this variability, and the result is video “jerkiness” (frame loss), audio degradation, and the like, when there is insufficient bandwidth to represent a frame at a given moment in time. Anyone who has participated in a videoconferencing session has experienced the uneven quality of conventional approaches.